- 25 7月, 2018 1 次提交
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由 Stephen Hemminger 提交于
Several files have extra line at end of file. Signed-off-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 24 5月, 2018 2 次提交
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由 Roopa Prabhu 提交于
This is a followup to fib rules sport, dport and ipproto match support. Only supports tcp, udp and icmp for ipproto. Used by fib rule self tests. Signed-off-by: NRoopa Prabhu <roopa@cumulusnetworks.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Alexei Starovoitov 提交于
bpfilter.ko consists of bpfilter_kern.c (normal kernel module code) and user mode helper code that is embedded into bpfilter.ko The steps to build bpfilter.ko are the following: - main.c is compiled by HOSTCC into the bpfilter_umh elf executable file - with quite a bit of objcopy and Makefile magic the bpfilter_umh elf file is converted into bpfilter_umh.o object file with _binary_net_bpfilter_bpfilter_umh_start and _end symbols Example: $ nm ./bld_x64/net/bpfilter/bpfilter_umh.o 0000000000004cf8 T _binary_net_bpfilter_bpfilter_umh_end 0000000000004cf8 A _binary_net_bpfilter_bpfilter_umh_size 0000000000000000 T _binary_net_bpfilter_bpfilter_umh_start - bpfilter_umh.o and bpfilter_kern.o are linked together into bpfilter.ko bpfilter_kern.c is a normal kernel module code that calls the fork_usermode_blob() helper to execute part of its own data as a user mode process. Notice that _binary_net_bpfilter_bpfilter_umh_start - end is placed into .init.rodata section, so it's freed as soon as __init function of bpfilter.ko is finished. As part of __init the bpfilter.ko does first request/reply action via two unix pipe provided by fork_usermode_blob() helper to make sure that umh is healthy. If not it will kill it via pid. Later bpfilter_process_sockopt() will be called from bpfilter hooks in get/setsockopt() to pass iptable commands into umh via bpfilter.ko If admin does 'rmmod bpfilter' the __exit code bpfilter.ko will kill umh as well. Signed-off-by: NAlexei Starovoitov <ast@kernel.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 18 4月, 2018 1 次提交
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由 David Ahern 提交于
Move logic of fib_convert_metrics into ip_metrics_convert. This allows the code that converts netlink attributes into metrics struct to be re-used in a later patch by IPv6. This is mostly a code move with the following changes to variable names: - fi->fib_net becomes net - fc_mx and fc_mx_len are passed as inputs pulled from fib_config - metrics array is passed as an input from fi->fib_metrics->metrics Signed-off-by: NDavid Ahern <dsahern@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 02 3月, 2018 1 次提交
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由 Yuval Mintz 提交于
The two implementations have almost identical structures - vif_device and mif_device. As a step toward uniforming the mr_tables, eliminate the mif_device and relocate the vif_device definition into a new common header file. Also, introduce a common initializing function for setting most of the vif_device fields in a new common source file. This requires modifying the ipv{4,6] Kconfig and ipv4 makefile as we're introducing a new common config option - CONFIG_IP_MROUTE_COMMON. Signed-off-by: NYuval Mintz <yuvalm@mellanox.com> Acked-by: NNikolay Aleksandrov <nikolay@cumulusnetworks.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 1月, 2018 1 次提交
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由 Masami Hiramatsu 提交于
Remove TCP probe module since jprobe has been deprecated. That function is now replaced by tcp/tcp_probe trace-event. You can use it via ftrace or perftools. Signed-off-by: NMasami Hiramatsu <mhiramat@kernel.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 02 11月, 2017 1 次提交
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由 Greg Kroah-Hartman 提交于
Many source files in the tree are missing licensing information, which makes it harder for compliance tools to determine the correct license. By default all files without license information are under the default license of the kernel, which is GPL version 2. Update the files which contain no license information with the 'GPL-2.0' SPDX license identifier. The SPDX identifier is a legally binding shorthand, which can be used instead of the full boiler plate text. This patch is based on work done by Thomas Gleixner and Kate Stewart and Philippe Ombredanne. How this work was done: Patches were generated and checked against linux-4.14-rc6 for a subset of the use cases: - file had no licensing information it it. - file was a */uapi/* one with no licensing information in it, - file was a */uapi/* one with existing licensing information, Further patches will be generated in subsequent months to fix up cases where non-standard license headers were used, and references to license had to be inferred by heuristics based on keywords. The analysis to determine which SPDX License Identifier to be applied to a file was done in a spreadsheet of side by side results from of the output of two independent scanners (ScanCode & Windriver) producing SPDX tag:value files created by Philippe Ombredanne. Philippe prepared the base worksheet, and did an initial spot review of a few 1000 files. The 4.13 kernel was the starting point of the analysis with 60,537 files assessed. Kate Stewart did a file by file comparison of the scanner results in the spreadsheet to determine which SPDX license identifier(s) to be applied to the file. She confirmed any determination that was not immediately clear with lawyers working with the Linux Foundation. Criteria used to select files for SPDX license identifier tagging was: - Files considered eligible had to be source code files. - Make and config files were included as candidates if they contained >5 lines of source - File already had some variant of a license header in it (even if <5 lines). All documentation files were explicitly excluded. The following heuristics were used to determine which SPDX license identifiers to apply. - when both scanners couldn't find any license traces, file was considered to have no license information in it, and the top level COPYING file license applied. For non */uapi/* files that summary was: SPDX license identifier # files ---------------------------------------------------|------- GPL-2.0 11139 and resulted in the first patch in this series. If that file was a */uapi/* path one, it was "GPL-2.0 WITH Linux-syscall-note" otherwise it was "GPL-2.0". Results of that was: SPDX license identifier # files ---------------------------------------------------|------- GPL-2.0 WITH Linux-syscall-note 930 and resulted in the second patch in this series. - if a file had some form of licensing information in it, and was one of the */uapi/* ones, it was denoted with the Linux-syscall-note if any GPL family license was found in the file or had no licensing in it (per prior point). Results summary: SPDX license identifier # files ---------------------------------------------------|------ GPL-2.0 WITH Linux-syscall-note 270 GPL-2.0+ WITH Linux-syscall-note 169 ((GPL-2.0 WITH Linux-syscall-note) OR BSD-2-Clause) 21 ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) 17 LGPL-2.1+ WITH Linux-syscall-note 15 GPL-1.0+ WITH Linux-syscall-note 14 ((GPL-2.0+ WITH Linux-syscall-note) OR BSD-3-Clause) 5 LGPL-2.0+ WITH Linux-syscall-note 4 LGPL-2.1 WITH Linux-syscall-note 3 ((GPL-2.0 WITH Linux-syscall-note) OR MIT) 3 ((GPL-2.0 WITH Linux-syscall-note) AND MIT) 1 and that resulted in the third patch in this series. - when the two scanners agreed on the detected license(s), that became the concluded license(s). - when there was disagreement between the two scanners (one detected a license but the other didn't, or they both detected different licenses) a manual inspection of the file occurred. - In most cases a manual inspection of the information in the file resulted in a clear resolution of the license that should apply (and which scanner probably needed to revisit its heuristics). - When it was not immediately clear, the license identifier was confirmed with lawyers working with the Linux Foundation. - If there was any question as to the appropriate license identifier, the file was flagged for further research and to be revisited later in time. In total, over 70 hours of logged manual review was done on the spreadsheet to determine the SPDX license identifiers to apply to the source files by Kate, Philippe, Thomas and, in some cases, confirmation by lawyers working with the Linux Foundation. Kate also obtained a third independent scan of the 4.13 code base from FOSSology, and compared selected files where the other two scanners disagreed against that SPDX file, to see if there was new insights. The Windriver scanner is based on an older version of FOSSology in part, so they are related. Thomas did random spot checks in about 500 files from the spreadsheets for the uapi headers and agreed with SPDX license identifier in the files he inspected. For the non-uapi files Thomas did random spot checks in about 15000 files. In initial set of patches against 4.14-rc6, 3 files were found to have copy/paste license identifier errors, and have been fixed to reflect the correct identifier. Additionally Philippe spent 10 hours this week doing a detailed manual inspection and review of the 12,461 patched files from the initial patch version early this week with: - a full scancode scan run, collecting the matched texts, detected license ids and scores - reviewing anything where there was a license detected (about 500+ files) to ensure that the applied SPDX license was correct - reviewing anything where there was no detection but the patch license was not GPL-2.0 WITH Linux-syscall-note to ensure that the applied SPDX license was correct This produced a worksheet with 20 files needing minor correction. This worksheet was then exported into 3 different .csv files for the different types of files to be modified. These .csv files were then reviewed by Greg. Thomas wrote a script to parse the csv files and add the proper SPDX tag to the file, in the format that the file expected. This script was further refined by Greg based on the output to detect more types of files automatically and to distinguish between header and source .c files (which need different comment types.) Finally Greg ran the script using the .csv files to generate the patches. Reviewed-by: NKate Stewart <kstewart@linuxfoundation.org> Reviewed-by: NPhilippe Ombredanne <pombredanne@nexb.com> Reviewed-by: NThomas Gleixner <tglx@linutronix.de> Signed-off-by: NGreg Kroah-Hartman <gregkh@linuxfoundation.org>
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- 16 6月, 2017 1 次提交
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由 Dave Watson 提交于
Add the infrustructure for attaching Upper Layer Protocols (ULPs) over TCP sockets. Based on a similar infrastructure in tcp_cong. The idea is that any ULP can add its own logic by changing the TCP proto_ops structure to its own methods. Example usage: setsockopt(sock, SOL_TCP, TCP_ULP, "tls", sizeof("tls")); modules will call: tcp_register_ulp(&tcp_tls_ulp_ops); to register/unregister their ulp, with an init function and name. A list of registered ulps will be returned by tcp_get_available_ulp, which is hooked up to /proc. Example: $ cat /proc/sys/net/ipv4/tcp_available_ulp tls There is currently no functionality to remove or chain ULPs, but it should be possible to add these in the future if needed. Signed-off-by: NBoris Pismenny <borisp@mellanox.com> Signed-off-by: NDave Watson <davejwatson@fb.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 3月, 2017 1 次提交
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由 Ido Schimmel 提交于
Most of the code concerned with the FIB notification chain currently resides in fib_trie.c, but this isn't really appropriate, as the FIB notification chain is also used for FIB rules. Therefore, it makes sense to move the common FIB notification code to a separate file and have it export the relevant functions, which can be invoked by its different users (e.g., fib_trie.c, fib_rules.c). Signed-off-by: NIdo Schimmel <idosch@mellanox.com> Signed-off-by: NJiri Pirko <jiri@mellanox.com> Acked-by: NDavid Ahern <dsa@cumulusnetworks.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 15 2月, 2017 1 次提交
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由 Steffen Klassert 提交于
This patch adds GRO ifrastructure and callbacks for ESP on ipv4 and ipv6. In case the GRO layer detects an ESP packet, the esp{4,6}_gro_receive() function does a xfrm state lookup and calls the xfrm input layer if it finds a matching state. The packet will be decapsulated and reinjected it into layer 2. Signed-off-by: NSteffen Klassert <steffen.klassert@secunet.com>
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- 24 10月, 2016 1 次提交
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由 Cyrill Gorcunov 提交于
In criu we are actively using diag interface to collect sockets present in the system when dumping applications. And while for unix, tcp, udp[lite], packet, netlink it works as expected, the raw sockets do not have. Thus add it. v2: - add missing sock_put calls in raw_diag_dump_one (by eric.dumazet@) - implement @destroy for diag requests (by dsa@) v3: - add export of raw_abort for IPv6 (by dsa@) - pass net-admin flag into inet_sk_diag_fill due to changes in net-next branch (by dsa@) v4: - use @pad in struct inet_diag_req_v2 for raw socket protocol specification: raw module carries sockets which may have custom protocol passed from socket() syscall and sole @sdiag_protocol is not enough to match underlied ones - start reporting protocol specifed in socket() call when sockets are raw ones for the same reason: user space tools like ss may parse this attribute and use it for socket matching v5 (by eric.dumazet@): - use sock_hold in raw_sock_get instead of atomic_inc, we're holding (raw_v4_hashinfo|raw_v6_hashinfo)->lock when looking up so counter won't be zero here. v6: - use sdiag_raw_protocol() helper which will access @pad structure used for raw sockets protocol specification: we can't simply rename this member without breaking uapi v7: - sine sdiag_raw_protocol() helper is not suitable for uapi lets rather make an alias structure with proper names. __check_inet_diag_req_raw helper will catch if any of structure unintentionally changed. CC: David S. Miller <davem@davemloft.net> CC: Eric Dumazet <eric.dumazet@gmail.com> CC: David Ahern <dsa@cumulusnetworks.com> CC: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru> CC: James Morris <jmorris@namei.org> CC: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org> CC: Patrick McHardy <kaber@trash.net> CC: Andrey Vagin <avagin@openvz.org> CC: Stephen Hemminger <stephen@networkplumber.org> Signed-off-by: NCyrill Gorcunov <gorcunov@openvz.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 9月, 2016 2 次提交
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由 Neal Cardwell 提交于
This commit implements a new TCP congestion control algorithm: BBR (Bottleneck Bandwidth and RTT). A detailed description of BBR will be published in ACM Queue, Vol. 14 No. 5, September-October 2016, as "BBR: Congestion-Based Congestion Control". BBR has significantly increased throughput and reduced latency for connections on Google's internal backbone networks and google.com and YouTube Web servers. BBR requires only changes on the sender side, not in the network or the receiver side. Thus it can be incrementally deployed on today's Internet, or in datacenters. The Internet has predominantly used loss-based congestion control (largely Reno or CUBIC) since the 1980s, relying on packet loss as the signal to slow down. While this worked well for many years, loss-based congestion control is unfortunately out-dated in today's networks. On today's Internet, loss-based congestion control causes the infamous bufferbloat problem, often causing seconds of needless queuing delay, since it fills the bloated buffers in many last-mile links. On today's high-speed long-haul links using commodity switches with shallow buffers, loss-based congestion control has abysmal throughput because it over-reacts to losses caused by transient traffic bursts. In 1981 Kleinrock and Gale showed that the optimal operating point for a network maximizes delivered bandwidth while minimizing delay and loss, not only for single connections but for the network as a whole. Finding that optimal operating point has been elusive, since any single network measurement is ambiguous: network measurements are the result of both bandwidth and propagation delay, and those two cannot be measured simultaneously. While it is impossible to disambiguate any single bandwidth or RTT measurement, a connection's behavior over time tells a clearer story. BBR uses a measurement strategy designed to resolve this ambiguity. It combines these measurements with a robust servo loop using recent control systems advances to implement a distributed congestion control algorithm that reacts to actual congestion, not packet loss or transient queue delay, and is designed to converge with high probability to a point near the optimal operating point. In a nutshell, BBR creates an explicit model of the network pipe by sequentially probing the bottleneck bandwidth and RTT. On the arrival of each ACK, BBR derives the current delivery rate of the last round trip, and feeds it through a windowed max-filter to estimate the bottleneck bandwidth. Conversely it uses a windowed min-filter to estimate the round trip propagation delay. The max-filtered bandwidth and min-filtered RTT estimates form BBR's model of the network pipe. Using its model, BBR sets control parameters to govern sending behavior. The primary control is the pacing rate: BBR applies a gain multiplier to transmit faster or slower than the observed bottleneck bandwidth. The conventional congestion window (cwnd) is now the secondary control; the cwnd is set to a small multiple of the estimated BDP (bandwidth-delay product) in order to allow full utilization and bandwidth probing while bounding the potential amount of queue at the bottleneck. When a BBR connection starts, it enters STARTUP mode and applies a high gain to perform an exponential search to quickly probe the bottleneck bandwidth (doubling its sending rate each round trip, like slow start). However, instead of continuing until it fills up the buffer (i.e. a loss), or until delay or ACK spacing reaches some threshold (like Hystart), it uses its model of the pipe to estimate when that pipe is full: it estimates the pipe is full when it notices the estimated bandwidth has stopped growing. At that point it exits STARTUP and enters DRAIN mode, where it reduces its pacing rate to drain the queue it estimates it has created. Then BBR enters steady state. In steady state, PROBE_BW mode cycles between first pacing faster to probe for more bandwidth, then pacing slower to drain any queue that created if no more bandwidth was available, and then cruising at the estimated bandwidth to utilize the pipe without creating excess queue. Occasionally, on an as-needed basis, it sends significantly slower to probe for RTT (PROBE_RTT mode). BBR has been fully deployed on Google's wide-area backbone networks and we're experimenting with BBR on Google.com and YouTube on a global scale. Replacing CUBIC with BBR has resulted in significant improvements in network latency and application (RPC, browser, and video) metrics. For more details please refer to our upcoming ACM Queue publication. Example performance results, to illustrate the difference between BBR and CUBIC: Resilience to random loss (e.g. from shallow buffers): Consider a netperf TCP_STREAM test lasting 30 secs on an emulated path with a 10Gbps bottleneck, 100ms RTT, and 1% packet loss rate. CUBIC gets 3.27 Mbps, and BBR gets 9150 Mbps (2798x higher). Low latency with the bloated buffers common in today's last-mile links: Consider a netperf TCP_STREAM test lasting 120 secs on an emulated path with a 10Mbps bottleneck, 40ms RTT, and 1000-packet bottleneck buffer. Both fully utilize the bottleneck bandwidth, but BBR achieves this with a median RTT 25x lower (43 ms instead of 1.09 secs). Our long-term goal is to improve the congestion control algorithms used on the Internet. We are hopeful that BBR can help advance the efforts toward this goal, and motivate the community to do further research. Test results, performance evaluations, feedback, and BBR-related discussions are very welcome in the public e-mail list for BBR: https://groups.google.com/forum/#!forum/bbr-dev NOTE: BBR *must* be used with the fq qdisc ("man tc-fq") with pacing enabled, since pacing is integral to the BBR design and implementation. BBR without pacing would not function properly, and may incur unnecessary high packet loss rates. Signed-off-by: NVan Jacobson <vanj@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNandita Dukkipati <nanditad@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: NVan Jacobson <vanj@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNandita Dukkipati <nanditad@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 6月, 2016 1 次提交
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由 Lawrence Brakmo 提交于
TCP-NV (New Vegas) is a major update to TCP-Vegas. An earlier version of NV was presented at 2010's LPC. It is a delayed based congestion avoidance for the data center. This version has been tested within a 10G rack where the HW RTTs are 20-50us and with 1 to 400 flows. A description of TCP-NV, including implementation details as well as experimental results, can be found at: http://www.brakmo.org/networking/tcp-nv/TCPNV.htmlSigned-off-by: NLawrence Brakmo <brakmo@fb.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 18 2月, 2016 1 次提交
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由 Ben Hutchings 提交于
There are no longer any in-tree drivers that use it. Signed-off-by: NBen Hutchings <ben@decadent.org.uk> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 1月, 2016 2 次提交
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由 Vladimir Davydov 提交于
tcp_memcontrol.c only contains legacy memory.tcp.kmem.* file definitions and mem_cgroup->tcp_mem init/destroy stuff. This doesn't belong to network subsys. Let's move it to memcontrol.c. This also allows us to reuse generic code for handling legacy memcg files. Signed-off-by: NVladimir Davydov <vdavydov@virtuozzo.com> Acked-by: NJohannes Weiner <hannes@cmpxchg.org> Cc: "David S. Miller" <davem@davemloft.net> Acked-by: NMichal Hocko <mhocko@suse.com> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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由 Johannes Weiner 提交于
Let the user know that CONFIG_MEMCG_KMEM does not apply to the cgroup2 interface. This also makes legacy-only code sections stand out better. [arnd@arndb.de: mm: memcontrol: only manage socket pressure for CONFIG_INET] Signed-off-by: NJohannes Weiner <hannes@cmpxchg.org> Cc: Michal Hocko <mhocko@suse.cz> Cc: Tejun Heo <tj@kernel.org> Acked-by: NVladimir Davydov <vdavydov@virtuozzo.com> Signed-off-by: NArnd Bergmann <arnd@arndb.de> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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- 21 10月, 2015 1 次提交
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由 Yuchung Cheng 提交于
This patch is the first half of the RACK loss recovery. RACK loss recovery uses the notion of time instead of packet sequence (FACK) or counts (dupthresh). It's inspired by the previous FACK heuristic in tcp_mark_lost_retrans(): when a limited transmit (new data packet) is sacked, then current retransmitted sequence below the newly sacked sequence must been lost, since at least one round trip time has elapsed. But it has several limitations: 1) can't detect tail drops since it depends on limited transmit 2) is disabled upon reordering (assumes no reordering) 3) only enabled in fast recovery ut not timeout recovery RACK (Recently ACK) addresses these limitations with the notion of time instead: a packet P1 is lost if a later packet P2 is s/acked, as at least one round trip has passed. Since RACK cares about the time sequence instead of the data sequence of packets, it can detect tail drops when later retransmission is s/acked while FACK or dupthresh can't. For reordering RACK uses a dynamically adjusted reordering window ("reo_wnd") to reduce false positives on ever (small) degree of reordering. This patch implements tcp_advanced_rack() which tracks the most recent transmission time among the packets that have been delivered (ACKed or SACKed) in tp->rack.mstamp. This timestamp is the key to determine which packet has been lost. Consider an example that the sender sends six packets: T1: P1 (lost) T2: P2 T3: P3 T4: P4 T100: sack of P2. rack.mstamp = T2 T101: retransmit P1 T102: sack of P2,P3,P4. rack.mstamp = T4 T205: ACK of P4 since the hole is repaired. rack.mstamp = T101 We need to be careful about spurious retransmission because it may falsely advance tp->rack.mstamp by an RTT or an RTO, causing RACK to falsely mark all packets lost, just like a spurious timeout. We identify spurious retransmission by the ACK's TS echo value. If TS option is not applicable but the retransmission is acknowledged less than min-RTT ago, it is likely to be spurious. We refrain from using the transmission time of these spurious retransmissions. The second half is implemented in the next patch that marks packet lost using RACK timestamp. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 28 8月, 2015 1 次提交
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由 Pravin B Shelar 提交于
geneve_core module handles send and receive functionality. This way OVS could use the Geneve API. Now with use of tunnel meatadata mode OVS can directly use Geneve netdevice. So there is no need for separate module for Geneve. Following patch consolidates Geneve protocol processing in single module. Signed-off-by: NPravin B Shelar <pshelar@nicira.com> Reviewed-by: NJesse Gross <jesse@nicira.com> Acked-by: NJohn W. Linville <linville@tuxdriver.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 6月, 2015 1 次提交
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由 Kenneth Klette Jonassen 提交于
CAIA Delay-Gradient (CDG) is a TCP congestion control that modifies the TCP sender in order to [1]: o Use the delay gradient as a congestion signal. o Back off with an average probability that is independent of the RTT. o Coexist with flows that use loss-based congestion control, i.e., flows that are unresponsive to the delay signal. o Tolerate packet loss unrelated to congestion. (Disabled by default.) Its FreeBSD implementation was presented for the ICCRG in July 2012; slides are available at http://www.ietf.org/proceedings/84/iccrg.html Running the experiment scenarios in [1] suggests that our implementation achieves more goodput compared with FreeBSD 10.0 senders, although it also causes more queueing delay for a given backoff factor. The loss tolerance heuristic is disabled by default due to safety concerns for its use in the Internet [2, p. 45-46]. We use a variant of the Hybrid Slow start algorithm in tcp_cubic to reduce the probability of slow start overshoot. [1] D.A. Hayes and G. Armitage. "Revisiting TCP congestion control using delay gradients." In Networking 2011, pages 328-341. Springer, 2011. [2] K.K. Jonassen. "Implementing CAIA Delay-Gradient in Linux." MSc thesis. Department of Informatics, University of Oslo, 2015. Cc: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Stephen Hemminger <stephen@networkplumber.org> Cc: Neal Cardwell <ncardwell@google.com> Cc: David Hayes <davihay@ifi.uio.no> Cc: Andreas Petlund <apetlund@simula.no> Cc: Dave Taht <dave.taht@bufferbloat.net> Cc: Nicolas Kuhn <nicolas.kuhn@telecom-bretagne.eu> Signed-off-by: NKenneth Klette Jonassen <kennetkl@ifi.uio.no> Acked-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 14 5月, 2015 1 次提交
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由 John W. Linville 提交于
net/ipv4/geneve.c -> net/ipv4/geneve_core.c This name better reflects the purpose of the module. Signed-off-by: NJohn W. Linville <linville@tuxdriver.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 10月, 2014 1 次提交
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由 Andy Zhou 提交于
This adds a device level support for Geneve -- Generic Network Virtualization Encapsulation. The protocol is documented at http://tools.ietf.org/html/draft-gross-geneve-01 Only protocol layer Geneve support is provided by this driver. Openvswitch can be used for configuring, set up and tear down functional Geneve tunnels. Signed-off-by: NJesse Gross <jesse@nicira.com> Signed-off-by: NAndy Zhou <azhou@nicira.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 29 9月, 2014 1 次提交
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由 Daniel Borkmann 提交于
This work adds the DataCenter TCP (DCTCP) congestion control algorithm [1], which has been first published at SIGCOMM 2010 [2], resp. follow-up analysis at SIGMETRICS 2011 [3] (and also, more recently as an informational IETF draft available at [4]). DCTCP is an enhancement to the TCP congestion control algorithm for data center networks. Typical data center workloads are i.e. i) partition/aggregate (queries; bursty, delay sensitive), ii) short messages e.g. 50KB-1MB (for coordination and control state; delay sensitive), and iii) large flows e.g. 1MB-100MB (data update; throughput sensitive). DCTCP has therefore been designed for such environments to provide/achieve the following three requirements: * High burst tolerance (incast due to partition/aggregate) * Low latency (short flows, queries) * High throughput (continuous data updates, large file transfers) with commodity, shallow buffered switches The basic idea of its design consists of two fundamentals: i) on the switch side, packets are being marked when its internal queue length > threshold K (K is chosen so that a large enough headroom for marked traffic is still available in the switch queue); ii) the sender/host side maintains a moving average of the fraction of marked packets, so each RTT, F is being updated as follows: F := X / Y, where X is # of marked ACKs, Y is total # of ACKs alpha := (1 - g) * alpha + g * F, where g is a smoothing constant The resulting alpha (iow: probability that switch queue is congested) is then being used in order to adaptively decrease the congestion window W: W := (1 - (alpha / 2)) * W The means for receiving marked packets resp. marking them on switch side in DCTCP is the use of ECN. RFC3168 describes a mechanism for using Explicit Congestion Notification from the switch for early detection of congestion, rather than waiting for segment loss to occur. However, this method only detects the presence of congestion, not the *extent*. In the presence of mild congestion, it reduces the TCP congestion window too aggressively and unnecessarily affects the throughput of long flows [4]. DCTCP, as mentioned, enhances Explicit Congestion Notification (ECN) processing to estimate the fraction of bytes that encounter congestion, rather than simply detecting that some congestion has occurred. DCTCP then scales the TCP congestion window based on this estimate [4], thus it can derive multibit feedback from the information present in the single-bit sequence of marks in its control law. And thus act in *proportion* to the extent of congestion, not its *presence*. Switches therefore set the Congestion Experienced (CE) codepoint in packets when internal queue lengths exceed threshold K. Resulting, DCTCP delivers the same or better throughput than normal TCP, while using 90% less buffer space. It was found in [2] that DCTCP enables the applications to handle 10x the current background traffic, without impacting foreground traffic. Moreover, a 10x increase in foreground traffic did not cause any timeouts, and thus largely eliminates TCP incast collapse problems. The algorithm itself has already seen deployments in large production data centers since then. We did a long-term stress-test and analysis in a data center, short summary of our TCP incast tests with iperf compared to cubic: This test measured DCTCP throughput and latency and compared it with CUBIC throughput and latency for an incast scenario. In this test, 19 senders sent at maximum rate to a single receiver. The receiver simply ran iperf -s. The senders ran iperf -c <receiver> -t 30. All senders started simultaneously (using local clocks synchronized by ntp). This test was repeated multiple times. Below shows the results from a single test. Other tests are similar. (DCTCP results were extremely consistent, CUBIC results show some variance induced by the TCP timeouts that CUBIC encountered.) For this test, we report statistics on the number of TCP timeouts, flow throughput, and traffic latency. 1) Timeouts (total over all flows, and per flow summaries): CUBIC DCTCP Total 3227 25 Mean 169.842 1.316 Median 183 1 Max 207 5 Min 123 0 Stddev 28.991 1.600 Timeout data is taken by measuring the net change in netstat -s "other TCP timeouts" reported. As a result, the timeout measurements above are not restricted to the test traffic, and we believe that it is likely that all of the "DCTCP timeouts" are actually timeouts for non-test traffic. We report them nevertheless. CUBIC will also include some non-test timeouts, but they are drawfed by bona fide test traffic timeouts for CUBIC. Clearly DCTCP does an excellent job of preventing TCP timeouts. DCTCP reduces timeouts by at least two orders of magnitude and may well have eliminated them in this scenario. 2) Throughput (per flow in Mbps): CUBIC DCTCP Mean 521.684 521.895 Median 464 523 Max 776 527 Min 403 519 Stddev 105.891 2.601 Fairness 0.962 0.999 Throughput data was simply the average throughput for each flow reported by iperf. By avoiding TCP timeouts, DCTCP is able to achieve much better per-flow results. In CUBIC, many flows experience TCP timeouts which makes flow throughput unpredictable and unfair. DCTCP, on the other hand, provides very clean predictable throughput without incurring TCP timeouts. Thus, the standard deviation of CUBIC throughput is dramatically higher than the standard deviation of DCTCP throughput. Mean throughput is nearly identical because even though cubic flows suffer TCP timeouts, other flows will step in and fill the unused bandwidth. Note that this test is something of a best case scenario for incast under CUBIC: it allows other flows to fill in for flows experiencing a timeout. Under situations where the receiver is issuing requests and then waiting for all flows to complete, flows cannot fill in for timed out flows and throughput will drop dramatically. 3) Latency (in ms): CUBIC DCTCP Mean 4.0088 0.04219 Median 4.055 0.0395 Max 4.2 0.085 Min 3.32 0.028 Stddev 0.1666 0.01064 Latency for each protocol was computed by running "ping -i 0.2 <receiver>" from a single sender to the receiver during the incast test. For DCTCP, "ping -Q 0x6 -i 0.2 <receiver>" was used to ensure that traffic traversed the DCTCP queue and was not dropped when the queue size was greater than the marking threshold. The summary statistics above are over all ping metrics measured between the single sender, receiver pair. The latency results for this test show a dramatic difference between CUBIC and DCTCP. CUBIC intentionally overflows the switch buffer which incurs the maximum queue latency (more buffer memory will lead to high latency.) DCTCP, on the other hand, deliberately attempts to keep queue occupancy low. The result is a two orders of magnitude reduction of latency with DCTCP - even with a switch with relatively little RAM. Switches with larger amounts of RAM will incur increasing amounts of latency for CUBIC, but not for DCTCP. 4) Convergence and stability test: This test measured the time that DCTCP took to fairly redistribute bandwidth when a new flow commences. It also measured DCTCP's ability to remain stable at a fair bandwidth distribution. DCTCP is compared with CUBIC for this test. At the commencement of this test, a single flow is sending at maximum rate (near 10 Gbps) to a single receiver. One second after that first flow commences, a new flow from a distinct server begins sending to the same receiver as the first flow. After the second flow has sent data for 10 seconds, the second flow is terminated. The first flow sends for an additional second. Ideally, the bandwidth would be evenly shared as soon as the second flow starts, and recover as soon as it stops. The results of this test are shown below. Note that the flow bandwidth for the two flows was measured near the same time, but not simultaneously. DCTCP performs nearly perfectly within the measurement limitations of this test: bandwidth is quickly distributed fairly between the two flows, remains stable throughout the duration of the test, and recovers quickly. CUBIC, in contrast, is slow to divide the bandwidth fairly, and has trouble remaining stable. CUBIC DCTCP Seconds Flow 1 Flow 2 Seconds Flow 1 Flow 2 0 9.93 0 0 9.92 0 0.5 9.87 0 0.5 9.86 0 1 8.73 2.25 1 6.46 4.88 1.5 7.29 2.8 1.5 4.9 4.99 2 6.96 3.1 2 4.92 4.94 2.5 6.67 3.34 2.5 4.93 5 3 6.39 3.57 3 4.92 4.99 3.5 6.24 3.75 3.5 4.94 4.74 4 6 3.94 4 5.34 4.71 4.5 5.88 4.09 4.5 4.99 4.97 5 5.27 4.98 5 4.83 5.01 5.5 4.93 5.04 5.5 4.89 4.99 6 4.9 4.99 6 4.92 5.04 6.5 4.93 5.1 6.5 4.91 4.97 7 4.28 5.8 7 4.97 4.97 7.5 4.62 4.91 7.5 4.99 4.82 8 5.05 4.45 8 5.16 4.76 8.5 5.93 4.09 8.5 4.94 4.98 9 5.73 4.2 9 4.92 5.02 9.5 5.62 4.32 9.5 4.87 5.03 10 6.12 3.2 10 4.91 5.01 10.5 6.91 3.11 10.5 4.87 5.04 11 8.48 0 11 8.49 4.94 11.5 9.87 0 11.5 9.9 0 SYN/ACK ECT test: This test demonstrates the importance of ECT on SYN and SYN-ACK packets by measuring the connection probability in the presence of competing flows for a DCTCP connection attempt *without* ECT in the SYN packet. The test was repeated five times for each number of competing flows. Competing Flows 1 | 2 | 4 | 8 | 16 ------------------------------ Mean Connection Probability 1 | 0.67 | 0.45 | 0.28 | 0 Median Connection Probability 1 | 0.65 | 0.45 | 0.25 | 0 As the number of competing flows moves beyond 1, the connection probability drops rapidly. Enabling DCTCP with this patch requires the following steps: DCTCP must be running both on the sender and receiver side in your data center, i.e.: sysctl -w net.ipv4.tcp_congestion_control=dctcp Also, ECN functionality must be enabled on all switches in your data center for DCTCP to work. The default ECN marking threshold (K) heuristic on the switch for DCTCP is e.g., 20 packets (30KB) at 1Gbps, and 65 packets (~100KB) at 10Gbps (K > 1/7 * C * RTT, [4]). In above tests, for each switch port, traffic was segregated into two queues. For any packet with a DSCP of 0x01 - or equivalently a TOS of 0x04 - the packet was placed into the DCTCP queue. All other packets were placed into the default drop-tail queue. For the DCTCP queue, RED/ECN marking was enabled, here, with a marking threshold of 75 KB. More details however, we refer you to the paper [2] under section 3). There are no code changes required to applications running in user space. DCTCP has been implemented in full *isolation* of the rest of the TCP code as its own congestion control module, so that it can run without a need to expose code to the core of the TCP stack, and thus nothing changes for non-DCTCP users. Changes in the CA framework code are minimal, and DCTCP algorithm operates on mechanisms that are already available in most Silicon. The gain (dctcp_shift_g) is currently a fixed constant (1/16) from the paper, but we leave the option that it can be chosen carefully to a different value by the user. In case DCTCP is being used and ECN support on peer site is off, DCTCP falls back after 3WHS to operate in normal TCP Reno mode. ss {-4,-6} -t -i diag interface: ... dctcp wscale:7,7 rto:203 rtt:2.349/0.026 mss:1448 cwnd:2054 ssthresh:1102 ce_state 0 alpha 15 ab_ecn 0 ab_tot 735584 send 10129.2Mbps pacing_rate 20254.1Mbps unacked:1822 retrans:0/15 reordering:101 rcv_space:29200 ... dctcp-reno wscale:7,7 rto:201 rtt:0.711/1.327 ato:40 mss:1448 cwnd:10 ssthresh:1102 fallback_mode send 162.9Mbps pacing_rate 325.5Mbps rcv_rtt:1.5 rcv_space:29200 More information about DCTCP can be found in [1-4]. [1] http://simula.stanford.edu/~alizade/Site/DCTCP.html [2] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp-final.pdf [3] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp_analysis-full.pdf [4] http://tools.ietf.org/html/draft-bensley-tcpm-dctcp-00 Joint work with Florian Westphal and Glenn Judd. Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com> Acked-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 20 9月, 2014 1 次提交
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由 Tom Herbert 提交于
This patch provides a receive path for foo-over-udp. This allows direct encapsulation of IP protocols over UDP. The bound destination port is used to map to an IP protocol, and the XFRM framework (udp_encap_rcv) is used to receive encapsulated packets. Upon reception, the encapsulation header is logically removed (pointer to transport header is advanced) and the packet is reinjected into the receive path with the IP protocol indicated by the mapping. Netlink is used to configure FOU ports. The configuration information includes the port number to bind to and the IP protocol corresponding to that port. This should support GRE/UDP (http://tools.ietf.org/html/draft-yong-tsvwg-gre-in-udp-encap-02), as will as the other IP tunneling protocols (IPIP, SIT). Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 15 7月, 2014 1 次提交
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由 Tom Herbert 提交于
Added udp_tunnel.c which can contain some common functions for UDP tunnels. The first function in this is udp_sock_create which is used to open the listener port for a UDP tunnel. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 25 2月, 2014 1 次提交
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由 Steffen Klassert 提交于
This patch add an IPsec protocol multiplexer. With this it is possible to add alternative protocol handlers as needed for IPsec virtual tunnel interfaces. Signed-off-by: NSteffen Klassert <steffen.klassert@secunet.com>
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- 07 1月, 2014 1 次提交
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由 Eric Dumazet 提交于
GRO/GSO layers can be enabled on a node, even if said node is only forwarding packets. This patch permits GSO (and upcoming GRO) support for GRE encapsulated packets, even if the host has no GRE tunnel setup. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: H.K. Jerry Chu <hkchu@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 04 7月, 2013 1 次提交
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由 Daniel Borkmann 提交于
Similarly to TCP/UDP offloading, move all related GRE functions to gre_offload.c to make things more explicit and similar to the rest of the code. Suggested-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 20 6月, 2013 1 次提交
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由 Pravin B Shelar 提交于
Refactor various ip tunnels xmit functions and extend iptunnel_xmit() so that there is more code sharing. Signed-off-by: NPravin B Shelar <pshelar@nicira.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 12 6月, 2013 1 次提交
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由 Daniel Borkmann 提交于
Similarly to TCP offloading and UDPv6 offloading, move all related UDPv4 functions to udp_offload.c to make things more explicit. Also, by this, we can make those functions static. Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 08 6月, 2013 1 次提交
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由 Daniel Borkmann 提交于
Would be good to make things explicit and move those functions to a new file called tcp_offload.c, thus make this similar to tcpv6_offload.c. While moving all related functions into tcp_offload.c, we can also make some of them static, since they are only used there. Also, add an explicit registration function. Suggested-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 27 3月, 2013 1 次提交
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由 Pravin B Shelar 提交于
Following patch refactors GRE code into ip tunneling code and GRE specific code. Common tunneling code is moved to ip_tunnel module. ip_tunnel module is written as generic library which can be used by different tunneling implementations. ip_tunnel module contains following components: - packet xmit and rcv generic code. xmit flow looks like (gre_xmit/ipip_xmit)->ip_tunnel_xmit->ip_local_out. - hash table of all devices. - lookup for tunnel devices. - control plane operations like device create, destroy, ioctl, netlink operations code. - registration for tunneling modules, like gre, ipip etc. - define single pcpu_tstats dev->tstats. - struct tnl_ptk_info added to pass parsed tunnel packet parameters. ipip.h header is renamed to ip_tunnel.h Signed-off-by: NPravin B Shelar <pshelar@nicira.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 8月, 2012 1 次提交
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由 Andrew Morton 提交于
Sanity: CONFIG_CGROUP_MEM_RES_CTLR -> CONFIG_MEMCG CONFIG_CGROUP_MEM_RES_CTLR_SWAP -> CONFIG_MEMCG_SWAP CONFIG_CGROUP_MEM_RES_CTLR_SWAP_ENABLED -> CONFIG_MEMCG_SWAP_ENABLED CONFIG_CGROUP_MEM_RES_CTLR_KMEM -> CONFIG_MEMCG_KMEM [mhocko@suse.cz: fix missed bits] Cc: Glauber Costa <glommer@parallels.com> Acked-by: NMichal Hocko <mhocko@suse.cz> Cc: Johannes Weiner <hannes@cmpxchg.org> Cc: KAMEZAWA Hiroyuki <kamezawa.hiroyu@jp.fujitsu.com> Cc: Hugh Dickins <hughd@google.com> Cc: Tejun Heo <tj@kernel.org> Cc: Aneesh Kumar K.V <aneesh.kumar@linux.vnet.ibm.com> Cc: David Rientjes <rientjes@google.com> Cc: KOSAKI Motohiro <kosaki.motohiro@jp.fujitsu.com> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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- 20 7月, 2012 1 次提交
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由 Yuchung Cheng 提交于
This patch impelements the common code for both the client and server. 1. TCP Fast Open option processing. Since Fast Open does not have an option number assigned by IANA yet, it shares the experiment option code 254 by implementing draft-ietf-tcpm-experimental-options with a 16 bits magic number 0xF989. This enables global experiments without clashing the scarce(2) experimental options available for TCP. When the draft status becomes standard (maybe), the client should switch to the new option number assigned while the server supports both numbers for transistion. 2. The new sysctl tcp_fastopen 3. A place holder init function Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 19 7月, 2012 1 次提交
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由 Saurabh 提交于
New VTI tunnel kernel module, Kconfig and Makefile changes. Signed-off-by: NSaurabh Mohan <saurabh.mohan@vyatta.com> Reviewed-by: NStephen Hemminger <shemminger@vyatta.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 7月, 2012 1 次提交
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由 David S. Miller 提交于
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 13 12月, 2011 1 次提交
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由 Glauber Costa 提交于
This patch introduces memory pressure controls for the tcp protocol. It uses the generic socket memory pressure code introduced in earlier patches, and fills in the necessary data in cg_proto struct. Signed-off-by: NGlauber Costa <glommer@parallels.com> Reviewed-by: NKAMEZAWA Hiroyuki <kamezawa.hiroyu@jp.fujtisu.com> CC: Eric W. Biederman <ebiederm@xmission.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 10 12月, 2011 1 次提交
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由 Pavel Emelyanov 提交于
Copy-s/tcp/udp/-paste from TCP bits. Signed-off-by: NPavel Emelyanov <xemul@parallels.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 14 5月, 2011 1 次提交
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由 Vasiliy Kulikov 提交于
This patch adds IPPROTO_ICMP socket kind. It makes it possible to send ICMP_ECHO messages and receive the corresponding ICMP_ECHOREPLY messages without any special privileges. In other words, the patch makes it possible to implement setuid-less and CAP_NET_RAW-less /bin/ping. In order not to increase the kernel's attack surface, the new functionality is disabled by default, but is enabled at bootup by supporting Linux distributions, optionally with restriction to a group or a group range (see below). Similar functionality is implemented in Mac OS X: http://www.manpagez.com/man/4/icmp/ A new ping socket is created with socket(PF_INET, SOCK_DGRAM, PROT_ICMP) Message identifiers (octets 4-5 of ICMP header) are interpreted as local ports. Addresses are stored in struct sockaddr_in. No port numbers are reserved for privileged processes, port 0 is reserved for API ("let the kernel pick a free number"). There is no notion of remote ports, remote port numbers provided by the user (e.g. in connect()) are ignored. Data sent and received include ICMP headers. This is deliberate to: 1) Avoid the need to transport headers values like sequence numbers by other means. 2) Make it easier to port existing programs using raw sockets. ICMP headers given to send() are checked and sanitized. The type must be ICMP_ECHO and the code must be zero (future extensions might relax this, see below). The id is set to the number (local port) of the socket, the checksum is always recomputed. ICMP reply packets received from the network are demultiplexed according to their id's, and are returned by recv() without any modifications. IP header information and ICMP errors of those packets may be obtained via ancillary data (IP_RECVTTL, IP_RETOPTS, and IP_RECVERR). ICMP source quenches and redirects are reported as fake errors via the error queue (IP_RECVERR); the next hop address for redirects is saved to ee_info (in network order). socket(2) is restricted to the group range specified in "/proc/sys/net/ipv4/ping_group_range". It is "1 0" by default, meaning that nobody (not even root) may create ping sockets. Setting it to "100 100" would grant permissions to the single group (to either make /sbin/ping g+s and owned by this group or to grant permissions to the "netadmins" group), "0 4294967295" would enable it for the world, "100 4294967295" would enable it for the users, but not daemons. The existing code might be (in the unlikely case anyone needs it) extended rather easily to handle other similar pairs of ICMP messages (Timestamp/Reply, Information Request/Reply, Address Mask Request/Reply etc.). Userspace ping util & patch for it: http://openwall.info/wiki/people/segoon/ping For Openwall GNU/*/Linux it was the last step on the road to the setuid-less distro. A revision of this patch (for RHEL5/OpenVZ kernels) is in use in Owl-current, such as in the 2011/03/12 LiveCD ISOs: http://mirrors.kernel.org/openwall/Owl/current/iso/ Initially this functionality was written by Pavel Kankovsky for Linux 2.4.32, but unfortunately it was never made public. All ping options (-b, -p, -Q, -R, -s, -t, -T, -M, -I), are tested with the patch. PATCH v3: - switched to flowi4. - minor changes to be consistent with raw sockets code. PATCH v2: - changed ping_debug() to pr_debug(). - removed CONFIG_IP_PING. - removed ping_seq_fops.owner field (unused for procfs). - switched to proc_net_fops_create(). - switched to %pK in seq_printf(). PATCH v1: - fixed checksumming bug. - CAP_NET_RAW may not create icmp sockets anymore. RFC v2: - minor cleanups. - introduced sysctl'able group range to restrict socket(2). Signed-off-by: NVasiliy Kulikov <segoon@openwall.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 02 2月, 2011 1 次提交
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由 David S. Miller 提交于
The time has finally come to remove the hash based routing table implementation in ipv4. FIB Trie is mature, well tested, and I've done an audit of it's code to confirm that it implements insert, delete, and lookup with the same identical semantics as fib_hash did. If there are any semantic differences found in fib_trie, we should simply fix them. I've placed the trie statistic config option under advanced router configuration. Signed-off-by: NDavid S. Miller <davem@davemloft.net> Acked-by: NStephen Hemminger <shemminger@vyatta.com>
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